• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.1
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• pp.374-395
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• 2017

Users quite often experience volatile channel conditions which negatively influence multimedia transmission. HTTP adaptive streaming has emerged as a new promising technology where the video quality can be adjusted to variable network conditions. Nevertheless, the new technology does not remain without drawbacks. As it has been observed, multiple video players sharing the same network link have often problems with achieving good efficiency and stability of play-out due to a mutual interference and competition among video players. Our investigation indicates that there may be another cause for under-performance of the streamed video. In an emulated environment, we implemented three algorithms of adaptive video play-out based on bandwidth or buffer assessment. As we show, traffic generated by players employing the same or similar play-out strategies is positively correlated and synchronised (clustered), whereas traffic originated from different play-out strategies shows negative or no correlations. However, when some of the parameters of the play-out strategies are randomised, the correlation and synchronisation diminish what has a positive impact on the smoothness of the traffic and on the video quality perceived by end users. Our research shows that non-correlated traffic flows generated by play-out strategies improve efficiency and stability of streamed adaptive video.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.549-551
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• 2004

This paper used OPNET to simulate video streaming service a test IP network and MPLS network with the traffic shaping that have with CQ_ LLQ algorithm, LSP of fixed bandwidth, policy of limitation users and measures parameters such as delay, throughput, packet loss. To verify the performance of video streaming service in IP network and MPLS network, two scenario that have same topology and traffic source. One is the simulation for best-effort service in pure IP network. The other is the simulation for QoS-enabled service in MPLS Network. Based on simulation result, the MPLS network with CQ_ LLQ algorithm and fixed LSP show advantage of the video streaming service QoS, specially delay and packet loss

• 한국정보통신설비학회:학술대회논문집
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• 2007.08a
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• pp.327-330
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• 2007

This article discusses a capacity planning method in QoS-guaranteed IP networks such as BcN (Broadband convergence Network). Since IP based networks have been developed to transport best-effort data traffic, the introduction of multi-service component in BcN requires fundamental modifications in capacity planning and network dimensioning. In this article, we present the key issues of the capacity planning in multi-service IP networks. To provide a foundation for network dimensioning procedure, we describe a systematic approach for classification and modeling of BcN traffic based on the QoS requirements of BcN services. We propose a capacity planning framework considering data traffic and real-time streaming traffic separately. The multi-service Erlang model, an extension of the conventional Erlang B loss model, is introduced to determine required link capacity for the call based real-time streaming traffic. The application of multi-service Erlang model can provide significant improvement in network planning due to sharing of network bandwidth among the different services.

• Journal of the Korea Society of Computer and Information
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• v.23 no.9
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• pp.35-42
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• 2018

In this paper, we propose an optimal streaming service method based on Hybrid CDN/P2P architecture. Recently, video streaming utilizes a CDN (Content Delivery Network) operation technique based on a Proxy Server, which is an end node located close to a user. However, since CDN has a fixed network traffic bandwidth and data information exchange among CDNs in the network is not smooth, it is difficult to guarantee traffic congestion and quality of image service. In the hybrid CDN/P2P network, a data selection technique is used to select only the data that is expected to be continuously requested among all the data in order to guarantee the QoS of the user who utilizes the limited bandwidth efficiently. In order to search user requested data, this technique effectively retrieves the storage information of the constituent nodes of CDN and P2P, and stores the new image information and calculates the deletion priority based on the request possibility as needed. Therefore, the streaming service scheme proposed in this paper can effectively improve the quality of the video streaming service on the network.

• The KIPS Transactions:PartC
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• v.18C no.6
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• pp.433-440
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• 2011

In this paper, we propose a new scheduling method that can guarantee reliable jitter by minimizing the queue length variation in the streaming service provisioning such as IPTV and VoD. The amount of traffic to be delivered within a certain time is very fluid because MPEG-4 and H.264 encoders use VBR(Variable Bit Rate) for delivering video streaming traffic. This VBR characteristic increases the end-to-end propagation delay variation when existing scheduling methods are used for delivering video frames. Therefore, we propose the new scheduling method that can minimize change rate of queue length by adaptively controling service rate taking into account the size of bulk incoming packets and arrival rate for bulk streaming traffic. Video frames can be more reliably transmitted through the minimization of the queue length variation using the proposed method. We use the queueing model and also carry out OPNET simulation to validate the proposed method.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.9
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• pp.3594-3610
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• 2015

Recent advancements in broadband wireless communication technologies enable mobile users to receive video streaming services with various smart devices. The long term evolution (LTE) network provides high bandwidth and low latency for several emerging mobile applications. This paper proposes the buffer aware scheduling (BAS) approach to schedule the downlink video traffic in LTE network. The proposed BAS scheme applies the weighting function to heuristically adjust the scheduling priority by considering the buffer status and channel condition of UE so as to reduce the time that UE stays in the connected state without receiving data. Both of 1080P and 2160P resolution video streaming sources were applied for exhaustive simulations to examine the performance of the proposed scheme by comparing to that of the fair bandwidth (FB) and the best channel quality indicator (CQI) schemes. The simulation results indicate that the proposed BAS scheme not only achieves better performance in power saving, streaming delivery time, and throughput than the FB scheme while maintaining the similar performance as the best CQI scheme in light traffic load. Specifically, the proposed scheme reduces streaming delivery time and generates less signaling overhead than the best CQI scheme when the traffic load is heavy.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4B
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• pp.227-233
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• 2007

According to the recent development of network technology, the demands of users are diversified and the needs of multimedia traffic are increasing. In general, UDP(User Datagram Protocol) traffic is used to transport multimedia data, which satisfied the real-time and isochronous characteristics. UDP traffic competes with TCP traffic and incur the network congestion. However, TCP traffic performs network congestion control but does not consider the receiver's status. Thus, it is not appropriate in case of streaming services. In this paper, we solve a fairness problems and proposed a network algorithm based on RTP/RTCP(Real-time Transport Protocol/Realtime Transport Control Protocol) in view of receiver status. The POBA(Priority Ordered Buffer Algorithm), which applies priorities in the receiver's buffer and networks, shows that it provides the appropriate environment for streaming services in view of packet loss ratio and buffer utilization of receiver's buffer compared with the previous method.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.313-320
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• 2005

The increasing popularity of multimedia streaming services introduces new challenges in content distribution. Especially, it is important to provide the QoS guarantees as they are increasingly expected to support the multimedia applications. Multimedia streams typically experience the high start-up delay due to the large protocol overhead, the delay, and the loss properties of the wireless networks. The service providers can improve the performance of multimedia streaming by caching the initial segment (prefix) of the popular streams at proxies near the requesting clients. The proxy can initiate transmission to the client while requesting the remainder of the stream from the server. In this paper, we propose the traffic based caching algorithm (TSLRU) to improve the performance of caching proxy. TSLRU classifies the traffic into three types, and improve the performance of caching proxy by reflecting the several elements such as traffic types, recency, frequency, object size when performing the replacement decision. In simulation, TSLRU performs better than the existing schemes in terms of byte hit rate, hit rate, startup latency, and throughput.

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.3
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• pp.328-335
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• 2002

When we want to transfer video streaming data through computer networks, we will have to be allocated a larger bandwidth compared to a general application. In general, it causes a serious network overload inevitably due to the limited bandwidth. In this paper, in order to resolve the problem, we haute taken a method for transmitting video streaming data using RTP and RTCP. With RR(Receiver Report) packet in RTCP we will test it to check whether the traffic in a network has occurred or not. If it happened, we haute tried to reduce the overall network traffic by dynamically changing the quantization factor of the Motion JPEG that is one of the encoding styles in JMF. When the ratio of the average of transmission for each session to the average of overall transmission is greater than 7%, we should adjust the amount of data to be transmitted for each session to reach the session mean values. The experimental results show that the proposed method taken here reduces the overload effectively and therefore improves the efficiency for transmitting video streaming data.

• Journal of KIISE:Information Networking
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• v.34 no.2
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• pp.100-109
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• 2007

Streaming protocol with the Available Bandwidth measurement scheme has problems that are to measure a Available Bandwidth uncorrectly and slowly. In this basis, in order to overcome limitations of the previous streaming protocols, we propose the NABO that is a New Available Bandwidth measurement scheme used by OWD(One-Way Delay). Proposed NABO(New Available Bandwidth measurement based on OWD) measures the constant transmission delay occurred by bottleneck link capacity and the variable delay. Competing traffic contribute to the variable delay. Through the measurement of the constant transmission delay and the competing traffic, a NABO can measure the Available Bandwidth correctly and fast in network. The simulation result proves that the proposed NABO has a performance that satisfies both accuracy viewpoint and measurement speed viewpoint.